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${ noResults }
53 Commits (16)
Author | SHA1 | Message | Date |
---|---|---|---|
|
14736ae6d4 |
res: Fix for Doxygen.
These are the remaining issues found in /res. ASTERISK-29761 Change-Id: I572e6019c422780dde5ce8448b6c85c77af6046d |
3 years ago |
|
923c836886 |
res: Spelling fixes
Correct typos of the following word families: identifying structures actcount initializer attributes statement enough locking declaration userevent provides unregister session execute searches verification suppressed prepared passwords recipients event because brief unidentified redundancy character the module reload operation backslashes accurate incorrect collision initializing instance interpreted buddies omitted manually requires queries generator scheduler configuration has owner resource performed masquerade apparently routable ASTERISK-29714 Change-Id: I88485116d2c59b776aa2e1f8b4ce8239a21decda |
3 years ago |
|
fd0ca1c3f9 |
Remove as much trailing whitespace as possible.
Change-Id: I873c1c6d00f447269bd841494459efccdd2c19c0 |
7 years ago |
|
83a2c4d2ae |
res_snmp: Declare RONLY if net-snmp headers do not.
Some net-snmp builds do not provide the RONLY declare, only NETSNMP_OLDAPI_RONLY. Map RONLY to NETSNMP_OLDAPI_RONLY to get around this error. Change-Id: Ida5c7ad9406515825485c4d3b4a34fd6ad0da577 |
8 years ago |
|
a6e5bae3ef |
Remove ASTERISK_REGISTER_FILE.
ASTERISK_REGISTER_FILE no longer has any purpose so this commit removes all traces of it. Previously exported symbols removed: * __ast_register_file * __ast_unregister_file * ast_complete_source_filename This also removes the mtx_prof static variable that was declared when MTX_PROFILE was enabled. This variable was only used in lock.c so it is now initialized in that file only. ASTERISK-26480 #close Change-Id: I1074af07d71f9e159c48ef36631aa432c86f9966 |
9 years ago |
|
478fb4a388 |
MALLOC_DEBUG: Replace WRAP_LIBC_MALLOC with ASTMM_LIBC.
There are 3 ways that calls directly to standard allocator functions can be dealt with: 1. Block their use, cause them to generate an error. This is the default. 2. Replace them with the Asterisk equivalent function calls. 3. Leave them alone. This change allows one of these 3 options to be selected by any source. The source just needs to define ASTMM_LIBC to ASTMM_BLOCK, ASTMM_REDIRECT, or ASTMM_IGNORE to use option 1, 2 or 3 respectively. Normally ASTMM_BLOCK is the correct option, so it is default when ASTMM_LIBC is not defined. In some cases when building 3rd party code it is desirable to have it use Asterisk functions, without changing the whole source - ASTMM_REDIRECT accomplishes this. When using 3rd party libraries sometimes a static inline function will make use of malloc or free. In these cases it may be unsafe to replace the allocator in the header, as it's possible the memory could be freed by the library using standard allocators. For those cases ASTMM_IGNORE is needed. Change-Id: I8afef4bc7f3b93914263ae27d3a5858b69663fc7 |
10 years ago |
|
4a58261694 |
git migration: Refactor the ASTERISK_FILE_VERSION macro
Git does not support the ability to replace a token with a version string during check-in. While it does have support for replacing a token on clone, this is somewhat sub-optimal: the token is replaced with the object hash, which is not particularly easy for human consumption. What's more, in practice, the source file version was often not terribly useful. Generally, when triaging bugs, the overall version of Asterisk is far more useful than an individual SVN version of a file. As a result, this patch removes Asterisk's support for showing source file versions. Specifically, it does the following: * Rename ASTERISK_FILE_VERSION macro to ASTERISK_REGISTER_FILE, and remove passing the version in with the macro. Other facilities than 'core show file version' make use of the file names, such as setting a debug level only on a specific file. As such, the act of registering source files with the Asterisk core still has use. The macro rename now reflects the new macro purpose. * main/asterisk: - Refactor the file_version structure to reflect that it no longer tracks a version field. - Remove the "core show file version" CLI command. Without the file version, it is no longer useful. - Remove the ast_file_version_find function. The file version is no longer tracked. - Rename ast_register_file_version/ast_unregister_file_version to ast_register_file/ast_unregister_file, respectively. * main/manager: Remove value from the Version key of the ModuleCheck Action. The actual key itself has not been removed, as doing so would absolutely constitute a backwards incompatible change. However, since the file version is no longer tracked, there is no need to attempt to include it in the Version key. * UPGRADE: Add notes for: - Modification to the ModuleCheck AMI Action - Removal of the "core show file version" CLI command Change-Id: I6cf0ff280e1668bf4957dc21f32a5ff43444a40e |
10 years ago |
|
e4738a59eb |
CHANNEL(peer), chan_iax2, res_fax, SNMP agent: Fix deadlock from reaching across a bridge.
Calling ast_channel_bridge_peer() cannot be done while holding any channel locks. The reported issue hit the deadlock in chan_iax2, but an audit of the ast_channel_bridge_peer() calls found three more locations where the same deadlock can occur. * Made CHANNEL(peer), res_fax, and the SNMP agent not call ast_channel_bridge_peer() with any channel locked. For CHANNEL(peer) I had to rework the logic to not hold the channel lock. * Made chan_iax2 no longer call ast_channel_bridge_peer(). It was done for legacy reasons that no longer apply. * Removed the iax.conf forcejitterbuffer option. It is now always enabled when the jitterbuffer option is enabled. If you put a jitter buffer on a channel it will be on the channel. ASTERISK-24600 #close Reported by: Jeff Collell Review: https://reviewboard.asterisk.org/r/4342/ ........ Merged revisions 430817 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430819 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
10 years ago |
|
dcf1ad14da |
Add module support level to ast_module_info structure. Print it in CLI "module show" .
ASTERISK-23919 #close Reported by Malcolm Davenport Review: https://reviewboard.asterisk.org/r/3802 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419592 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
|
11d2993426 |
Get the SNMP code to compile.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396126 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
d1f8e338b0 |
Add The Status Of A Module To The Output Of "CLI> module show"
When a module's configuration is not loadable, we still load the module but it is not in a running state. When trying to troubleshoot, let's say, why chan_motif is ignoring inbound XMPP traffic, there is no way to indicate that a loaded module is not currently running. (closes issue ASTERISK-21108) Reported by: Rusty Newton Tested by: Michael L. Young Patches: asterisk-21108_add_status-v2.diff Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2331/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381749 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
166b4e2b30 |
Multiple revisions 369001-369002
........ r369001 | kpfleming | 2012-06-15 10:56:08 -0500 (Fri, 15 Jun 2012) | 11 lines Add support-level indications to many more source files. Since we now have tools that scan through the source tree looking for files with specific support levels, we need to ensure that every file that is a component of a 'core' or 'extended' module (or the main Asterisk binary) is explicitly marked with its support level. This patch adds support-level indications to many more source files in tree, but avoids adding them to third-party libraries that are included in the tree and to source files that don't end up involved in Asterisk itself. ........ r369002 | kpfleming | 2012-06-15 10:57:14 -0500 (Fri, 15 Jun 2012) | 3 lines Add a script to enable finding source files without support-levels defined. ........ Merged revisions 369001-369002 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369005 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369013 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
6d6bacd5cb |
Convert some strncpys to ast_copy_string
Review: https://reviewboard.asterisk.org/r/1732/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362635 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
786f5898d1 |
Finalize ast_channel opaquification
Review: https://reviewboard.asterisk.org/r/1786/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358907 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
0e5c761c28 |
Opaquify ast_channel typedefs, fd arrays, and softhangup flag
Review: https://reviewboard.asterisk.org/r/1784/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357721 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
a9d607a357 |
Opaquify ast_channel structs and lists
Review: https://reviewboard.asterisk.org/r/1773/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357542 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
57f42bd74f |
ast_channel opaquification of pointers and integral types
Review: https://reviewboard.asterisk.org/r/1753/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356042 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
34c55e8e7c |
Opaquify char * and char[] in ast_channel
Review: https://reviewboard.asterisk.org/r/1733/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354968 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
27b69e7d29 |
Audit of ao2_iterator_init() usage for v1.8.
Fixes numerous reference leaks and missing ao2_iterator_destroy() calls as a result. Review: https://reviewboard.asterisk.org/r/1697/ ........ Merged revisions 352955 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352956 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352957 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
99cae5b750 |
Opaquify channel stringfields
Continue channel opaque-ification by wrapping all of the stringfields. Eventually, we will restrict what can actually set these variables, but the purpose for now is to hide the implementation and keep people from adding code that directly accesses the channel structure. Semantic changes will follow afterward. Review: https://reviewboard.asterisk.org/r/1661/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352348 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
04da92c379 |
Replace direct access to channel name with accessor functions
There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
cf7bbcc4c6 |
Expand the caller ANI field to an ast_party_id
Expand the ani field in ast_party_caller and ast_party_connected_line to an ast_party_id. This is an extension to the ast_callerid restructuring patch in review: https://reviewboard.asterisk.org/r/702/ Review: https://reviewboard.asterisk.org/r/744/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276393 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
2cf60bb09d |
Make compile again.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276391 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
ec37ffbdaf |
ast_callerid restructuring
The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
a5a0a5f867 |
Consolidate ast_channel.cid.cid_rdnis into ast_channel.redirecting.from.number.
SWP-1229 ABE-2161 * Ensure chan_local.c:local_call() will not leak cid.cid_dnid when copying. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256104 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
cdd1f9e296 |
Finish implementaton of astobj2 OBJ_MULTIPLE, and convert ast_channel_iterator to use it.
This patch finishes the implementation of OBJ_MULTIPLE in astobj2 (the case where multiple results need to be returned; OBJ_NODATA mode already was supported). In addition, it converts ast_channel_iterators (only the targeted versions, not the ones that iterate over all channels) to use this method. During this work, I removed the 'ao2_flags' arguments to the ast_channel_iterator constructor functions; there were no uses of that argument yet, there is only one possible flag to pass, and it made the iterators less 'opaque'. If at some point in the future someone really needs an ast_channel_iterator that does not lock the container, we can provide constructor(s) for that purpose. Review: https://reviewboard.asterisk.org/r/379/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225244 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
642bec4d6f |
AST-2009-005
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@211539 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
cba19c8a67 |
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big improvement for performance, stability, code maintainability, and ease of future code development. The channel list is no longer an unsorted linked list. The main container for channels is an astobj2 hash table. All of the code related to searching for channels or iterating active channels has been rewritten. Let n be the number of active channels. Iterating the channel list has gone from O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1). Searching for a channel by extension is still O(n), but uses a new method for doing so, which is more efficient. The ast_channel object is now a reference counted object. The benefits here are plentiful. Some benefits directly related to issues in the previous code include: 1) When threads other than the channel thread owning a channel wanted access to a channel, it had to hold the lock on it to ensure that it didn't go away. This is no longer a requirement. Holding a reference is sufficient. 2) There are places that now require less dealing with channel locks. 3) There are places where channel locks are held for much shorter periods of time. 4) There are places where dealing with more than one channel at a time becomes _MUCH_ easier. ChanSpy is a great example of this. Writing code in the future that deals with multiple channels will be much easier. Some additional information regarding channel locking and reference count handling can be found in channel.h, where a new section has been added that discusses some of the rules associated with it. Mark Michelson also assisted with the development of this patch. He did the conversion of ChanSpy and introduced a new API, ast_autochan, which makes it much easier to deal with holding on to a channel pointer for an extended period of time and having it get automatically updated if the channel gets masqueraded. Mark was also a huge help in the code review process. Thanks to David Vossel for his assistance with this branch, as well. David did the conversion of the DAHDIScan application by making it become a wrapper for ChanSpy internally. The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch. Review: http://reviewboard.digium.com/r/203/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
4ec301360c |
Merge a large set of updates to the Asterisk indications API.
This patch includes a number of changes to the indications API. The primary motivation for this work was to improve stability. The object management in this API was significantly flawed, and a number of trivial situations could cause crashes. The changes included are: 1) Remove the module res_indications. This included the critical functionality that actually loaded the indications configuration. I have seen many people have Asterisk problems because they accidentally did not have an indications.conf present and loaded. Now, this code is in the core, and Asterisk will fail to start without indications configuration. There was one part of res_indications, the dialplan applications, which did belong in a module, and have been moved to a new module, app_playtones. 2) Object management has been significantly changed. Tone zones are now managed using astobj2, and it is no longer possible to crash Asterisk by issuing a reload that destroys tone zones while they are in use. 3) The API documentation has been filled out. 4) The API has been updated to follow our naming conventions. 5) Various bits of code throughout the tree have been updated to account for the API update. 6) Configuration parsing has been mostly re-written. 7) "Code cleanup" The code is from svn/asterisk/team/russell/indications/. Review: http://reviewboard.digium.com/r/149/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176627 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
e77b3cea6b |
Merged revisions 174148 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r174148 | russell | 2009-02-07 10:15:07 -0600 (Sat, 07 Feb 2009) | 2 lines Fix a race condition that could cause a crash. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174149 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
ef6ad2b53c |
Merged revisions 168561 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168561 | russell | 2009-01-13 13:13:05 -0600 (Tue, 13 Jan 2009) | 2 lines Revert unnecessary indications API change from rev 122314 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168562 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
1a4a30aaea |
Use ast_str_strlen() instead of recalculating the string length.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164054 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
517afad041 |
nuke another use of the ast_str internals.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164028 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
ef965af270 |
Make res_snmp.so compile on OpenBSD.
OpenBSD uses an old version of gcc which throws an error if you use a macro that's not #defined git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162583 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
1235d47242 |
Make res/snmp/agent.c build
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115199 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
3c594ab9fa |
Merged revisions 105572 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105572 | qwell | 2008-03-03 12:06:52 -0600 (Mon, 03 Mar 2008) | 7 lines Fix types for astNumChannels and astConfigCallsProcessed. (closes issue #12114) Reported by: jeffg Patches: 12114.patch uploaded by jeffg (license 192) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105573 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
6bb33538e1 |
small change to allow this file to compile. No problem if you don't install the libsnmp package.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104301 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
4f0cb30f4a |
Convert this file over the new method of getting the Asterisk version.
(I don't have this building on this machine, so caio1982 on IRC is going to test it for me. :) ) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96743 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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5e24835995 |
modify http://svn.digium.com/view/asterisk?view=rev&rev=93603
so that paths and filename are writable by asterisk.c without causing segfaults. This involves defining the variables as const char *, and having them point to as static, writable buffer defined in asterisk.c On passing, fix some errors in using these variables in some files in utils/ , and in res/snmp/agent.c which was redefining a variable without using paths.h (not applicable to 1.4) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94168 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
094ec6521d |
make netsmp build under AST_DEVMODE. Description, included in the source,
is below. I should note that the PACKAGE_* macros that asterisk defines in autoconfig.h are not used anywhere in the tree so they should just be removed. /* * There is some collision collision between netsmp and asterisk names, * causing build under AST_DEVMODE to fail. * * The following PACKAGE_* macros are one place. * Also netsnmp has an improper check for HAVE_DMALLOC_H, using * #if HAVE_DMALLOC_H instead of #ifdef HAVE_DMALLOC_H * As a countermeasure we define it to 0, however this will fail * when the proper check is implemented. */ No git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93875 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
d5b454bf8d |
Convert ast_verbose to ast_verb.
Reported by: snuffy Patch by: snuffy (Closes issue #11547) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@92913 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
02ea9face7 |
Add count of total number of calls processed by asterisk during it's lifetime.
Add number of total calls and current calls to SNMP. Closes issue #10057, patch by jcmoore. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@91779 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
c50975fd06 |
Fix a typo in the Asterisk MIB and fix astNumChanBridged so it acts as a counter again
(closes issue #10118, patch by jeffg) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@80510 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
81bc1d7af5 |
Merge in ast_strftime branch, which changes timestamps to be accurate to the microsecond, instead of only to the second
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75706 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
f379df9773 |
Simplify some logic and convert spaces to tabs
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69068 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
5d350e0f81 |
The variable used for the return value must be declared as static. I broke
this when applying the patch, sorry! (issue #9637, jeffg) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69067 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
3b9a36109f |
Add support for retrieving the number of channels that are currently bridged
via SNMP. (closes issue #9637, initial patch from jeffg, modified by me) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@68507 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
09f75aa6dc |
rename the structs struct tone_zone_sound and struct tone_zone
defined in indications.h to ind_tone_zone_sound and ind_tone_zone, to avoid conflicts with the structs with the same names defined in tonezone.h Hope i haven't missed any instance. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48958 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
09de4d2b67 |
Update res_snmp to use new API declaration of pbx_builtin_serialize_variables (issue #8627 reported by johann8384)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48595 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
e3ddee7d7f |
make latest header file changes for this file too
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@33482 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |